Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker
RtpVideoStreamReceiver::receive_cs_ is not really necessary, since all of the functions where that lock is acquired, are arrived at from functions of BaseChannel which DCHECK being called from BaseChannel::worker_thread_.
BUG=webrtc:8037
Review-Url: https://codereview.webrtc.org/2987933003
Cr-Original-Commit-Position: refs/heads/master@{#19508}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 8b07305b048fb91840ae7f9a58c6df7115540015
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 6064e69..01cb4dc 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -314,37 +314,33 @@
// This method handles both regular RTP packets and packets recovered
// via FlexFEC.
void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
- // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
- // RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
- {
- rtc::CritScope lock(&receive_cs_);
- if (!receiving_) {
- return;
- }
+ if (!receiving_) {
+ return;
+ }
- if (!packet.recovered()) {
- int64_t now_ms = clock_->TimeInMilliseconds();
+ if (!packet.recovered()) {
+ int64_t now_ms = clock_->TimeInMilliseconds();
- // Periodically log the RTP header of incoming packets.
- if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
- std::stringstream ss;
- ss << "Packet received on SSRC: " << packet.Ssrc()
- << " with payload type: " << static_cast<int>(packet.PayloadType())
- << ", timestamp: " << packet.Timestamp()
- << ", sequence number: " << packet.SequenceNumber()
- << ", arrival time: " << packet.arrival_time_ms();
- int32_t time_offset;
- if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
- ss << ", toffset: " << time_offset;
- }
- uint32_t send_time;
- if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
- ss << ", abs send time: " << send_time;
- }
- LOG(LS_INFO) << ss.str();
- last_packet_log_ms_ = now_ms;
+ // Periodically log the RTP header of incoming packets.
+ if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
+ std::stringstream ss;
+ ss << "Packet received on SSRC: " << packet.Ssrc()
+ << " with payload type: " << static_cast<int>(packet.PayloadType())
+ << ", timestamp: " << packet.Timestamp()
+ << ", sequence number: " << packet.SequenceNumber()
+ << ", arrival time: " << packet.arrival_time_ms();
+ int32_t time_offset;
+ if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
+ ss << ", toffset: " << time_offset;
}
+ uint32_t send_time;
+ if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
+ ss << ", abs send time: " << send_time;
+ }
+ LOG(LS_INFO) << ss.str();
+ last_packet_log_ms_ = now_ms;
}
}
@@ -442,8 +438,7 @@
}
void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) {
- // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
- // RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
RTC_DCHECK(std::find(secondary_sinks_.cbegin(), secondary_sinks_.cend(),
sink) == secondary_sinks_.cend());
secondary_sinks_.push_back(sink);
@@ -451,8 +446,7 @@
void RtpVideoStreamReceiver::RemoveSecondarySink(
const RtpPacketSinkInterface* sink) {
- // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
- // RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
auto it = std::find(secondary_sinks_.begin(), secondary_sinks_.end(), sink);
if (it == secondary_sinks_.end()) {
// We might be rolling-back a call whose setup failed mid-way. In such a
@@ -486,6 +480,7 @@
void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
if (rtp_payload_registry_.IsRed(header)) {
int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
if (packet[header.headerLength] == ulpfec_pt) {
@@ -510,7 +505,6 @@
return;
if (packet_length > sizeof(restored_packet_))
return;
- rtc::CritScope lock(&receive_cs_);
if (restored_packet_in_use_) {
LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
return;
@@ -568,11 +562,10 @@
bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
- {
- rtc::CritScope lock(&receive_cs_);
- if (!receiving_) {
- return false;
- }
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
+
+ if (!receiving_) {
+ return false;
}
rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
@@ -634,12 +627,12 @@
}
void RtpVideoStreamReceiver::StartReceive() {
- rtc::CritScope lock(&receive_cs_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
receiving_ = true;
}
void RtpVideoStreamReceiver::StopReceive() {
- rtc::CritScope lock(&receive_cs_);
+ RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
receiving_ = false;
}
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 4d63d76..15af2e6 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -32,7 +32,7 @@
#include "webrtc/modules/video_coding/sequence_number_util.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/thread_checker.h"
+#include "webrtc/rtc_base/sequenced_task_checker.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -183,11 +183,11 @@
const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
- rtc::CriticalSection receive_cs_;
- bool receiving_ GUARDED_BY(receive_cs_);
- uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
- bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
- int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
+ rtc::SequencedTaskChecker worker_task_checker_;
+ bool receiving_ GUARDED_BY(worker_task_checker_);
+ uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(worker_task_checker_);
+ bool restored_packet_in_use_ GUARDED_BY(worker_task_checker_);
+ int64_t last_packet_log_ms_ GUARDED_BY(worker_task_checker_);
const std::unique_ptr<RtpRtcp> rtp_rtcp_;
@@ -210,10 +210,8 @@
bool has_received_frame_;
- // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
- // rtc::ThreadChecker worker_thread_checker_;
- std::vector<RtpPacketSinkInterface*> secondary_sinks_; // This needs
- // to be GUARDED_BY(worker_thread_checker_).
+ std::vector<RtpPacketSinkInterface*> secondary_sinks_
+ GUARDED_BY(worker_task_checker_);
};
} // namespace webrtc
